application notes for configuring avaya aura ... · the calls to cs1000e (see figure 1). when a sip...

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Solution & Interoperability Test Lab Application Notes

©2015 Avaya Inc. All Rights Reserved.

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Avaya Solution & Interoperability Test Lab

Application Notes for Configuring Avaya Aura ®

Communication Server 1000 R7.65, Avaya Aura ® Session

Manager 7.0 and Avaya Session Border Controller for

Enterprise R7.0 to support BT Global Services SIP Trunk

Platform (NOAS) - Issue 0.1

Abstract

These Application Notes describe the steps used to configure Session Initiation Protocol (SIP)

trunking between BT Global Services SIP Trunk and an Avaya SIP enabled Enterprise

Solution. The Avaya solution consists of Avaya Session Border Controller for Enterprise,

Avaya Aura® Session Manager and Avaya Communication Server 1000. BT is a member of

the DevConnect Service Provider program.

Information in these Application Notes has been obtained through DevConnect compliance

testing and additional technical discussions. Testing was conducted via the DevConnect

Program at the Avaya Solution and Interoperability Test Lab.

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1. Introduction These Application Notes describe the steps used to configure Session Initiation Protocol (SIP)

trunking between BT Global Services SIP Trunk and an Avaya SIP-enabled enterprise solution.

The Avaya solution consists of the following: Avaya Communication Server 1000 R7.65

(CS1000); Avaya Aura ® Session Manager R7.0 (Session Manager) and Avaya Session Border

Controller for Enterprise R7.0 (Avaya SBCE). Note that the shortened names shown in brackets

will be used throughout the remainder of the document. Customers using this Avaya SIP-enabled

enterprise solution with BT Global Services SIP Trunk are able to place and receive PSTN calls

via a dedicated Internet connection and the SIP protocol. This converged network solution is an

alternative to traditional PSTN trunks. This approach generally results in lower cost for the

enterprise customer.

2. General Test Approach and Test Results The general test approach was to configure a simulated enterprise site using an Avaya SIP

telephony solution consisting of Communication Server 10000, Session Manager and Avaya

SBCE. The enterprise site was configured to connect to the BT Global Services SIP Trunk

Platform.

DevConnect Compliance Testing is conducted jointly by Avaya and DevConnect members. The

jointly-defined test plan focuses on exercising APIs and/or standards-based interfaces pertinent

to the interoperability of the tested products and their functionalities. DevConnect Compliance

Testing is not intended to substitute full product performance or feature testing performed by

DevConnect members, nor is it to be construed as an endorsement by Avaya of the suitability or

completeness of a DevConnect member’s solution.

2.1. Interoperability Compliance Testing

The interoperability test included the following:

Incoming calls to the enterprise site from the PSTN were routed to the DID numbers

assigned by BT. Incoming PSTN calls were terminated on Digital, Unistim, SIP and

Analog telephones at the enterprise side.

Outgoing calls from the enterprise site were completed via BT to PSTN telephones.

Outgoing calls from the enterprise to the PSTN were made from Digital, Unistim, SIP

and Analog telephones.

Calls using the G.711A and G.729A codecs.

Fax calls to/from a group 3 fax machine to a PSTN connected fax machine using T.38.

DTMF transmission using RFC 2833 with successful Voice Mail/Vector navigation for

inbound and outbound calls.

User features such as hold and resume, transfer, conference, call forwarding, etc.

Caller ID Presentation and Caller ID Restriction.

Call coverage and call forwarding for endpoints at the enterprise site.

Transmission and response of SIP OPTIONS messages sent by BT Global Services SIP

Trunk Platform requiring Avaya response and sent by Avaya requiring BT response.

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2.2. Test Results

Interoperability testing of the sample configuration was completed with successful results for BT

Global Services SIP Trunk with the following observations:

The CS1000 default configuration will not allow a blind transfer to be executed (incoming

SIP Service Provider trunk to outgoing SIP Service Provider trunk) if the SIP Service

Provider in question does not support the SIP UPDATE method. With the installation of

plugin 501 on the CS1000, the blind transfer will be allowed and the call will be completed.

The limitation of this plugin is that no ringback is provided to the originator of the call for the

duration that the destination set is ringing. In addition to plugin 501, it is required that VTRK

SU version “cs1000-vtrk-7.65.16.22.-4.i386.000.ntl” or higher be used on all SSG

signalling servers to ensure proper operation of the blind transfer feature. The use of plugin

501 does not restrict the use of the SIP UPDATE method of blind transfer to other parties

that do happen to support the UPDATE method, but rather extends support to those parties

that do not. Note that plugin 501 is independent of and does not require the Global Plugin

Package 409.

When testing failover to alternative network SBC, outgoing calls took approximately 32

seconds to set up. A subsequent call did not attempt to set up via the non-operational

SBC and was established within an acceptable time though there was no audio. An

attempt was made to reduce the initial setup time by reducing SIP timer T1 on the Avaya

SBCE but this did not function according to RFC 3261. Fault Report AURORA-7344

was raised to have this investigated by the Avaya SBCE support team.

The SIP Trunk between the Avaya Galway Lab and the BT Sandbox was unstable and

became non-operational several times during testing. This was deemed to be a network

issue and not related to the functionality of the BT Global Services SIP Trunk Platform.

The network responded to an outbound call to an invalid PSTN number with 404

“Service Unavailable-No ports available”. This behaviour did not create an issue and a

tone was heard on the calling phone. It is noted however, as the commonly used response

is 404 “Not Found”.

The BT Sandbox did not have a voicemail system in operation at the time of test. Instead

DTMF was successfully tested using IVR.

There are no mobile phones available on the BT sandbox so Mobile X feature was tested

with a fixed phone.

All unwanted MIME was stripped on outbound calls using the Adaptation Module in

Session Manager.

No inbound toll free numbers were tested as none were available from the Service

Provider.

No Emergency Services numbers tested as test calls to these numbers should be pre-

arranged with the Operator.

2.3. Support

For technical support on BT Global Services products please contact BT Global Services on

0800 028 5314 or visit their website at www.globalservices.bt.com

http://www.globalservices.bt.com/

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3. Reference Configuration Figure 1 illustrates the test configuration. The test configuration shows an Enterprise site

connected to BT’s SIP Trunk Service. Located at the Enterprise site is an Avaya SBCE, Session

Manager and CS1000. Endpoints are Avaya 1140 series IP telephones (with Unistim and SIP

firmware), Avaya 1200 series IP telephones (with Unistim and SIP firmware), Avaya IP 2050PC

Softphone, Avaya Digital telephone, Analog telephone and fax machine. For security purposes,

any public IP addresses or PSTN routable phone numbers used in the compliance test are not

shown in these Application Notes.

Figure 1: Test Setup BT SIP Trunk to Avaya Enterprise

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4. Equipment and Software Validated The following equipment and software were used for the sample configuration provided:

Equipment/Software Release/Version

Avaya

Avaya Aura® Session Manager 7.0.0.0.700007

Avaya Aura® System Manager 7.0.0.0.16266

Avaya Aura® Communication Server 1000 Avaya Communication Server 1000E

R7.6

Version 7.65.P

Deplist: CPL_X21_07_65P

All CS1000E patches listed in Appendix

A

Avaya Communication Server 1000E Media

Gateway

CSP Version: MGCC DC01

MSP Version: MGCM AB02

APP Version: MGCA BA18

FPGA Version: MGCF AA22

BOOT Version: MGCB BA18

DSP1 Version: DSP2 AB07

Avaya Session Border Controller for

Enterprise

7.0.0-21-6602

Avaya 1140e and 1230 Unistim Telephones FW: 0625C8A

Avaya 1140e and 1230 SIP Telephones FW: 04.10.18.00.bin

Avaya 2050PC Release 4.3.0081

Avaya Analogue Telephone N/A

Avaya M3904 Digital Telephone N/A

BT Global Services

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5. Configure Avaya Communication Server 1000 This section describes the steps required to configure CS1000 for SIP Trunking and also the

basic configuration for telephones (analog, SIP and IP phones). SIP trunks are established

between CS1000 and Session Manager. SIP trunks are also established between Session

Manager and the Avaya SBCE private interface. The Avaya SBCE public interface connects to

the BT Global Services SIP trunks. Incoming PSTN calls from the BT Global Services SIP

Trunk service traverse the Avaya SBCE and are directed to the Session Manager, which directs

the calls to CS1000E (see Figure 1).

When a SIP message arrives at CS1000, further incoming call treatment, such as incoming digit

translations and class of service restrictions may be performed. All outgoing calls to the PSTN

are processed within CS1000 and may be first subject to outbound features such as route

selection, digit manipulation and class of service restrictions. When CS1000 selects a SIP trunk

for outgoing PSTN calls, SIP signaling is directed to Session Manager. Session Manager directs

the outbound SIP messages to the Avaya SBCE private interface. The Avaya SBCE public

interface manages outgoing SIP sessions onwards to the BT Global Services SIP trunks.

Specific CS1000 configuration was performed using Element Manager and the system terminal

interface. The general installation of the CS1000, System Manager, Session Manager and Avaya

SBCE is presumed to have been previously completed and is not discussed here. Configuration

details will be provided as required to draw attention to changes in default system configurations.

5.1. Logging into the Avaya Communication Server 1000E

Configuration on the CS1000 will be performed by using both SSH Putty session and Avaya

Unified Communications Management GUI.

Log in using SSH to the ELAN IP address of the Call Server with a username containing the

correct privileges. Once logged in type csconsole, this will take the user into the vxworks shell of

the call server. Next type login; the user will then be asked to login with correct credentials.

Once logged-in the user can then progress to load any overlay.

Log in using the web based Avaya Unified Communications Management GUI. Avaya Unified

Communications Management GUI may be launched directly via http://<ipaddress> where the

relevant <ipaddress> is the TLAN IP address of the CS1000. Avaya Unified Communications

Management can also be implemented on System Manager.

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The following screen shows the login screen. Login with the appropriate credentials.

The Avaya Unified Communications Management Elements page will be used for configuration.

Click on the Element Name corresponding to CS1000E in the Element Type column. In the

abridged screen below, the user would click on the Element Name EM on cs1kvl9.

5.2. Confirm System Features

The keycode installed on the Call Server controls the maximum values for these attributes. If a

required feature is not enabled or there is insufficient capacity, contact an authorized Avaya sales

representative to add additional capacity. Use the CS1000 system terminal and manually load

overlay 22 to print the System Limits (the required command is slt ), and verify that the number

of SIP Access Ports reported by the system is sufficient for the combination of trunks to the BT

Global Services network, and any other SIP trunks needed. See the following screenshot for a

typical System Limits printout. The value of SIP ACCESS PORTS defines the maximum

number of SIP trunks for the CS1000.

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System type is - Communication Server 1000/CP PM

CP PM - Pentium M 1.4 GHz

IPMGs Registered: 4

IPMGs Unregistered: 0

IPMGs Configured/unregistered: 2

TRADITIONAL TELEPHONES 120 LEFT 110 USED 10

DECT USERS 16 LEFT 16 USED 0

IP USERS 10000 LEFT 9954 USED 46

BASIC IP USERS 16 LEFT 13 USED 3

TEMPORARY IP USERS 8 LEFT 8 USED 0

DECT VISITOR USER 16 LEFT 16 USED 0

ACD AGENTS 192 LEFT 185 USED 7

MOBILE EXTENSIONS 8 LEFT 7 USED 1

TELEPHONY SERVICES 16 LEFT 13 USED 3

CONVERGED MOBILE USERS 8 LEFT 8 USED 0

AVAYA SIP LINES 16 LEFT 12 USED 4

THIRD PARTY SIP LINES 16 LEFT 16 USED 0

PCA 20 LEFT 18 USED 2

ITG ISDN TRUNKS 0 LEFT 0 USED 0

H.323 ACCESS PORTS 524 LEFT 524 USED 0

AST 6652 LEFT 6640 USED 12

SIP CONVERGED DESKTOPS 16 LEFT 16 USED 0

SIP CTI TR87 16 LEFT 8 USED 8

SIP ACCESS PORTS 524 LEFT 518 USED 6

RAN CON 90 LEFT 90 USED 0

MUS CON 120 LEFT 120 USED 0

Load Overlay 21 and confirm the customer is setup to use ISDN trunks by typing the PRT and

NET_DATA commands as shown below.

REQ: prt

TYPE: net

TYPE NET_DATA

CUST 0

TYPE NET_DATA

CUST 00

OPT RTD

AC1 INTL NPA SPN NXX LOC

AC2

FNP YES

ISDN YES

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5.3. Configure Codecs for Voice and FAX operation

BT Global Service’s SIP Trunk supports G.711A and G.729 voice codecs. Using the CS1000

Element Manager sidebar, select Nodes, Servers, Media Cards. Navigate to the IP Network

IP Telephony Nodes Node Details VGW and Codecs property page and configure the

CS1000 General codec settings as in the following screenshots. The values highlighted are

required for correct operation. The following screenshot shows the necessary General settings.

Move down to the Voice Codecs section and configure the G.711 codec settings. The following

screenshot shows the G.711 codec settings.

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Next, scroll down to the G.729 codec section and configure the settings.

Finally, configure the Fax settings as in the highlighted section of the next screenshot. Click on

the Save button when finished.

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5.4. Virtual Trunk Gateway Configuration

Use CS1000 Element Manager to configure the system node properties. Navigate to the System

IP Networks IP Telephony Nodes Node Details and verify the highlighted section is

completed with the correct IP addresses and subnet masks of the Node. The call server and

signaling server have previously been configured with IP addresses. The Node IPv4 address is

the IP address that the IP phones use to register. This is also where the SIP trunk connection is

made to Session Manager. When an entity link is added in Session Manager for the CS1000, it is

the Node IPv4 address that is used (see Section 6.5 – Define SIP Entities for more details).

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The next two screenshots show the SIP Virtual Trunk Gateway configuration, navigate to

System IP Networks IP Telephony Nodes Node Details Gateway (SIPGW)

Virtual Trunk Configuration Details and fill in the highlighted areas with the relevant settings.

Vtrk gateway application: Provides option to select Gateway applications. The three

supported modes are SIP Gateway (SIPGw), H.323Gw, and SIPGw and H.323Gw

SIP domain name: The SIP domain name is the SIP Service Domain. The SIP domain

name configured in the Signaling Server properties must match the Service Domain name

configured in Session Manager; in this case avaya.com

Local SIP port: The Local SIP Port is the port to which the gateway listens. The default

value is 5060

Gateway endpoint name: This field cannot be left blank so a value is needed here. This

field is used when a Network Routing Server is used for registration of the endpoint. In

this network a Session Manager is used so any value can be put in here and will not be

used

Application node ID: This is a unique value that can be alphanumeric and is for the new

Node that is being created, in this case 200

Proxy or Redirect Server: Primary TLAN IP address is the Security Module IP address

of Session Manager. The Transport protocol used for SIP, in this case is TCP

SIP URI Map: Public E.164 - National and Private - Unknown are left blank. All

other fields in the SIP URI Map are left with default values

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5.5. Configure Bandwidth Zones

Bandwidth Zones are used for alternate call routing between IP stations and for bandwidth

management. SIP trunks require a unique zone, not shared with other resources and best practice

dictates that IP telephones and Media Gateways are all placed in separate zones. In the sample

configuration SIP trunks use zone 01 and IP and SIP Telephones use zone 02; system defaults

were used for each zone other than the parameter configured for Zone Intent. For SIP Trunks

(zone 01), VTRK is configured for Zone Intent. For IP, SIP Telephones (zone 02), MO is

configured for Main Office.

Use Element Manager to define bandwidth zones as in the following highlighted example. Use

Element Manager and navigate to System IP Network Zones Bandwidth Zones and

add new zones as required.

5.6. Configure Incoming Digit Conversion Table

A limited number of Direct Dial Inwards (DDI) numbers were available. The Incoming Digit

Conversion (IDC) table was configured to translate incoming PSTN numbers to four digit local

telephone extension numbers. The digits of the actual PSTN DDI number are obscured for

security reasons. The following screenshot shows the incoming PSTN numbers converted to

local extension numbers. These were altered during testing to map to various SIP, Analog,

Digital or UNIStim telephones depending on the particular test case being executed.

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5.7. Configure SIP Trunks

CS1000 virtual trunks will be used for all inbound and outbound PSTN calls to the BT Global

Services SIP Trunk service. Six separate steps are required to configure CS1000 virtual trunks:

Configure a D-Channel Handler (DCH); configure using the CS1000 system terminal

and overlay 17

Configure a SIP trunk Route Data Block (RDB); configure using the CS1000 system

terminal and overlay 16

Configure SIP trunk members; configure using the CS1000 system terminal and overlay

14

Configure a Digit Manipulation Data Block (DGT), configure using the CS1000 system


Configure a Route List Block (RLB); configure using the CS1000 system terminal and

overlay 86

Configure Co-ordinated Dialling Plan(s) (CDP); configure using the CS1000 system


The following is an example DCH configuration for SIP trunks. Load Overlay 17 at the CS1000

system terminal and enter the following values. The highlighted entries are required for correct

SIP trunk operation. Exit overlay 17 when completed.

Overlay 17 ADAN DCH 1 CTYP DCIP DES VIR_TRK

USR ISLD

ISLM 4000

SSRC 3700

OTBF 32

NASA YES

IFC SL1 CNEG 1

RLS ID 4

RCAP ND2

MBGA NO

H323

OVLR NO

OVLS NO

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Next, configure the SIP trunk Route Data Block (RDB) using the CS1000 system terminal and

overlay 16. Load Overlay 16, enter RDB at the prompt, press return and commence

configuration. The value for DCH is the same as previously entered in overlay 17. The value for

NODE should match the node value in Section 5.4. The value for ZONE should match that used

in Section 5.5 for VTRK. The remaining highlighted values are important for correct SIP trunk

operation.

Overlay 16 TYPE: RDB

CUST 00

ROUT 1

TYPE RDB

CUST 00

ROUT 1 DES VIR_TRK

TKTP TIE NPID_TBL_NUM 0

ESN NO

RPA NO

CNVT NO

SAT NO

RCLS EXT

VTRK YES ZONE 00001 PCID SIP CRID NO

NODE 200 DTRK NO

ISDN YES MODE ISLD DCH 1 IFC SL1 PNI 00000

NCNA YES

NCRD YES

TRO NO

FALT NO

CTYP UKWN

INAC NO

ISAR NO

DAPC NO

MBXR NO

MBXOT NPA

MBXT 0

PTYP ATT

CNDP UKWN

AUTO NO

DNIS NO

DCDR NO

ICOG IAO SRCH LIN

TRMB YES

STEP

ACOD 1111 TCPP NO

PII NO

AUXP NO

TARG

CLEN 1

BILN NO

OABS

INST

IDC YES DCNO 0

NDNO 0 *

DEXT NO

DNAM NO

SIGO STD

STYP SDAT

MFC NO

ICIS YES

OGIS YES

TIMR ICF 1920

OGF 1920

EOD 13952

LCT 256

DSI 34944

NRD 10112

DDL 70

ODT 4096

RGV 640

GTO 896

GTI 896

SFB 3

PRPS 800

NBS 2048

NBL 4096

IENB 5

TFD 0

VSS 0

VGD 6

EESD 1024

SST 5 0

DTD NO

SCDT NO

2 DT NO

NEDC ORG

FEDC ORG

CPDC NO

DLTN NO

HOLD 02 02 40

SEIZ 02 02

SVFL 02 02

DRNG NO

CDR NO

NATL YES

SSL

CFWR NO

IDOP NO

VRAT NO

MUS YES

MRT 21

PANS YES

RACD NO

MANO NO

FRL 0 0

FRL 1 0

FRL 2 0

FRL 3 0

FRL 4 0

FRL 5 0

FRL 6 0

FRL 7 0

OHQ NO

OHQT 00

CBQ NO

AUTH NO

TTBL 0

ATAN NO

OHTD NO

PLEV 2

OPR NO

ALRM NO

ART 0

PECL NO

DCTI 0

TIDY 1600 100

ATRR NO

TRRL NO

SGRP 0

ARDN NO

CTBL 0

AACR NO

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Next, configure virtual trunk members using the CS1000 system terminal and Overlay 14.

Configure sufficient trunk members to carry both incoming and outgoing PSTN calls. The

following example shows a single SIP trunk member configuration. Load Overlay 14 at the

system terminal and type new X, where X is the required number of trunks. Continue entering

data until the overlay exits. The RTMB value is a combination of the ROUT value entered in the

previous step and the first trunk member (usually 1). The remaining highlighted values are

important for correct SIP trunk operation.

Overlay 14 TN 100 0 0 0

DATE

PAGE

DES VIR_TRK

TN 100 0 00 00 VIRTUAL

TYPE IPTI CDEN 8D

CUST 0

XTRK VTRK ZONE 00001 TIMP 600

BIMP 600

AUTO_BIMP NO

NMUS NO

TRK ANLG

NCOS 0

RTMB 1 1 CHID 1

TGAR 1

STRI/STRO IMM IMM SUPN YES

AST NO

IAPG 0

CLS UNR DIP CND ECD WTA LPR APN THFD XREP SPCD MSBT

P10 NTC

TKID

AACR NO

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Next, configure a Digit Manipulation data block (DGT) in overlay 86. Load Overlay 86 at the

system terminal and type new. The following example shows the values used. Note: ISPN is set

to 0 as BT Global Services required a prefix of 0 to be inserted before the dialed number for

outbound calls. The value for Digit Manipulation Index (DMI) is the same as when inputting the

DMI value during configuration of the Route List Block.

Overlay 86 CUST 0

FEAT dgt

DMI 10 DEL 0

ISPN 0

CTYP NPA

Configure a Route List Block (RLB) in overlay 86. Load Overlay 86 at the system terminal and

type new. The following example shows the values used. The value for ROUT is the same as

previously entered in overlay 16. The RLI value is unique to each RLB.

Overlay 86 CUST 0

FEAT rlb

RLI 10 ELC NO

ENTR 0

LTER NO

ROUT 1 TOD 0 ON 1 ON 2 ON 3 ON

4 ON 5 ON 6 ON 7 ON

VNS NO

SCNV NO

CNV NO

EXP NO

FRL 0

DMI 10

CTBL 0

ISDM 0

FCI 0

FSNI 0

BNE NO

DORG NO

SBOC NRR

PROU 1

IDBB DBD

IOHQ NO

OHQ NO

CBQ NO

ISET 0

NALT 5

MFRL 0

OVLL 0

Next, configure Co-ordinated Dialling Plan(s) (CDP) which users will dial to reach PSTN

numbers. Use the CS1000 system terminal and Overlay 87. The following are some example

CDP entries used. The highlighted RLI value previously configured in overlay 86 is used as the

Route List Index (RLI), this is the default PSTN route to the SIP Trunk service.

TSC 00353

FLEN 0

RRPA NO

RLI 10 CCBA NO

TSC 18

FLEN 0

RRPA NO

RLI 10 CCBA NO

TSC 800

FLEN 0

RRPA NO

RLI 10 CCBA NO

TSC 08

FLEN 0

RRPA NO

RLI 10 CCBA NO

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5.8. Calling Line Identification

This section documents basic configuration relevant to the BT Global Services configuration.

Load Overlay 15 at system terminal and enter the required values in bold. As shown below,

CLID is set to YES and ENTRY is set to 0. HNTN and HLCL match the required digits

assigned by BT Global Services and DIDN is set to NO.

Load Overlay 15

TYPE NET_DATA

CUST 0

OPT

AC2

FNP

CLID YES

SIZE

INTL

ENTRY 0

HNTN 07689

ESA_HLCL

ESA_INHN NO

ESA_APDN NO

HLCL 11010

DIDN NO

DIDN_LEN 0

HLOC

LSC

CLASS_FMT DN

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5.9. Configure Analog, Digital and IP Telephones

A variety of telephone types were used during the testing, the following is the configuration for

the Avaya 1140e UNIStim IP telephone. Load Overlay 20 at the system terminal and enter the

following values. A unique four digit number is entered for the KEY 00. The value for

CFG_ZONE is the value used in Section 5.5 for IP and SIP Telephones.

Load Overlay 20 IP Telephone configuration DES 1140


TYPE 1140

CDEN 8D

CTYP XDLC

CUST 0

NUID

NHTN

CFG_ZONE 00002 CUR_ZONE 00002

ERL 0

ECL 0

FDN 0

TGAR 0

LDN NO

NCOS 0

SGRP 0

RNPG 1

SCI 0

SSU

LNRS 16

XLST

SCPW

SFLT NO

CAC_MFC 0

CLS UNR FBA WTA LPR PUA MTD FNA HTA TDD HFA CRPD

MWA LMPN RMMD SMWD AAD IMD XHD IRD NID OLD VCE DRG1

POD SLKD CCSD SWD LNA CNDA

CFTD SFD MRD DDV CNID CDCA MSID DAPA BFED RCBD

ICDA CDMD LLCN MCTD CLBD AUTR

GPUD DPUD DNDA CFXA ARHD FITD CLTD ASCD

CPFA CPTA ABDD CFHD FICD NAID BUZZ AGRD MOAD

UDI RCC HBTA AHD IPND DDGA NAMA MIND PRSD NRWD NRCD NROD

DRDD EXR0

USMD USRD ULAD CCBD RTDD RBDD RBHD PGND OCBD FLXD FTTC DNDY DNO3 MCBN

FDSD NOVD VOLA VOUD CDMR PRED RECA MCDD T87D SBMD KEM3 MSNV FRA PKCH MUTA MWTD

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---continued from previous page----

DVLD CROD CROD

CPND_LANG ENG

RCO 0

HUNT 0

LHK 0

PLEV 02

PUID

DANI NO

AST 00

IAPG 1

AACS NO

ITNA NO

DGRP

MLWU_LANG 0

MLNG ENG

DNDR 0

KEY 00 MCR 6000 0 MARP CPND

CPND_LANG ROMAN

NAME IP1140

XPLN 10

DISPLAY_FMT FIRST,LAST

01 MCR 6000 0 CPND

CPND_LANG ROMAN

NAME IP1140

XPLN 10


02

03 BSY

04 DSP

05

06

07

08

09

10

11

12

13

14

15

16

17 TRN

18 AO6

19 CFW 16

20 RGA

21 PRK

22 RNP

23

24 PRS

25 CHG

26 CPN

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Digital telephones are configured using the overlay 20; the following is a sample 3904 digital set

configuration. Again, a unique number is entered for the KEY 00 and KEY 01 value.

Overlay 20 – Digital Set configuration

TYPE: 3904

DES 3904


TYPE 3904

CDEN 8D

CTYP XDLC

CUST 0

MRT

ERL 0

FDN 0

TGAR 0

LDN NO

NCOS 0

SGRP 0

RNPG 1

SCI 0

SSU

LNRS 16

XLST

SCPW

SFLT NO

CAC_MFC 0

CLS UNR FBD WTA LPR PUA MTD FND HTD TDD HFA GRLD CRPA STSD

MWA LMPN RMMD SMWD AAD IMD XHD IRD NID OLD VCE DRG1

POD SLKD CCSD SWD LNA CNDA


ICDA CDMA LLCN MCTD CLBD AUTU

GPUD DPUD DNDA CFXA ARHD FITD CNTD CLTD ASCD

CPFA CPTA ABDA CFHD FICD NAID BUZZ AGRD MOAD

UDI RCC HBTD AHA IPND DDGA NAMA MIND PRSD NRWD NRCD NROD

DRDD EXR0


FDSD NOVD CDMR PRED RECA MCDD T87D SBMD PKCH CROD CROD

CPND_LANG ENG

RCO 0

HUNT

PLEV 02

PUID

DANI NO

SPID NONE

AST

IAPG 1

AACS

ACQ

ASID

SFNB

SFRB

USFB

CALB

FCTB

ITNA NO

DGRP

PRI 01

MLWU_LANG 0

---continued on next page----

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---continued from previous page----

MLNG ENG

DNDR 0


CPND_LANG ROMAN

NAME Digital Set

XPLN 10


01 MCR 6066 0 CPND

CPND_LANG ROMAN

NAME Digital Set

XPLN 10


02 DSP

03 MSB

04

05

06

07

08

09

10

11

12

13

14

15

16

17 TRN

18 AO6

19 CFW 16

20 RGA

21 PRK

22 RNP

23

24 PRS

25 CHG

26 CPN

27 CLT

28 RLT

29

30

31

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Analog telephones are also configured using overlay 20; the following example shows an analog

port configured for Plain Ordinary Telephone Service (POTS) and also configured to allow fax

transmission. A unique value is entered for DN, this is the extension number. DTN is required if

the telephone uses DTMF dialing. Values FAXA and MPTD configure the port for T.38 Fax

transmissions.

Overlay 20 – Analog Telephone Configuration

DES 500

TN 100 0 00 03

TYPE 500

CDEN 4D

CUST 0

MRT

ERL 00000

WRLS NO

DN 6004

AST NO

IAPG 0

HUNT

TGAR 0

LDN NO

NCOS 0

SGRP 0

RNPG 0

XLST

SCI 0

SCPW

SFLT NO

CAC_MFC 0

CLS UNR DTN FBD XFD WTA THFD FND HTD ONS

LPR XRD AGRD CWD SWD MWD RMMD SMWD LPD XHD SLKD CCSD LND TVD

CFTD SFD MRD C6D CNID CLBD AUTU

ICDD CDMD LLCN EHTD MCTD

GPUD DPUD CFXD ARHD OVDD AGTD CLTD LDTD ASCD SDND

MBXD CPFA CPTA UDI RCC HBTD IRGD DDGA NAMA MIND

NRWD NRCD NROD SPKD CRD PRSD MCRD

EXR0 SHL SMSD ABDD CFHD DNDY DNO3

CWND USMD USRD CCBD BNRD OCBD RTDD RBDD RBHD FAXA CNUD CNAD PGND FTTC

FDSD NOVD CDMR PRED MCDD T87D SBMD PKCH MPTD

PLEV 02

PUID

AACS NO

MLWU_LANG 0

FTR DCFW 4

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5.10. Configure the SIP Line Gateway Service

SIP terminal operation requires the CS1000 node to be configured as a SIP Line Gateway (SLG)

before SIP telephones can be configured. Prior to configuring the SIP Line node properties, the

SIP Line service must be enabled in the customer data block. Use the CS1000 system terminal

and overlay 15 to activate SIP Line services (SLS_DATA), as in the following example where

SIPL_ON is set to YES.

SLS_DATA

SIPL_ON YES UAPR 11 NMME NO

If a numerical value is entered against the UAPR setting, this number will be pre appended to all

SIP Line configurations, and is used internally in the SIP Line server to track SIP terminals. Use

Element Manager and navigate to the IP Network IP Telephony Nodes Node Details

SIP Line Gateway Configuration page. See the following screenshot for highlighted critical

parameters.

SIP Line Gateway Application: Enable the SIP line service on the node, check the box

to enable

SIP Domain Name: The value must match that configured in Section 6.2

SLG endpoint name: The endpoint name is the same endpoint name as the SIP Line

Gateway and will be used for SIP gateway registration

SLG Local Sip port: Default value is 5070

SLG Local TLS port: Default value is 5071

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5.1. Configure SIP Line Telephones

When SIP Line service configuration is completed, use the CS1000 system terminal and Overlay

20 to add a Universal Extension (UEXT). See the following example of a SIP Line extension.

The value for UXTY must be SIPL. This example is for an Avaya SIP telephone, so the value

for SIPN is 1. The SIPU value is the username, SCPW is the logon password and these values

are required to register the SIP telephone to the SLG. The value for CFG_ZONE is the value

used in Section 5.5 for IP and SIP Telephones. A unique telephone number is entered for value

KEY 00. The value for KEY 01 is comprised of the UAPR (set in Section 5.8) value and the

telephone number used in KEY 00.

Load Overlay 20 – SIP Telephone Configuration DES SIPD


TYPE UEXT

CDEN 8D

CTYP XDLC

CUST 0

UXTY SIPL MCCL YES

SIPN 1 SIP3 0

FMCL 0

TLSV 0

SIPU 6002 NDID 200

SUPR NO

SUBR DFLT MWI RGA CWI MSB

UXID

NUID

NHTN

CFG_ZONE 00002 CUR_ZONE 00002

ERL 0

ECL 0

VSIT NO

FDN

TGAR 0

LDN NO

NCOS 0

SGRP 0

RNPG 0

SCI 0

SSU

XLST

SCPW 1234 SFLT NO

CAC_MFC 0

CLS UNR FBD WTA LPR MTD FNA HTA TDD HFD CRPD

MWD LMPN RMMD SMWD AAD IMD XHD IRD NID OLD VCE DRG1

POD SLKD CCSD SWD LND CNDA


ICDD CDMD LLCN MCTD CLBD AUTU

GPUD DPUD DNDA CFXA ARHD FITD CLTD ASCD

CPFA CPTA ABDD CFHD FICD NAID BUZZ AGRD MOAD

---continued on next page---

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---continued from previous page---

UDI RCC HBTD AHA IPND DDGA NAMA MIND PRSD NRWD NRCD NROD

DRDD EXR0


FDSD NOVD VOLA VOUD CDMR PRED RECD MCDD T87D SBMD ELMD MSNV FRA PKCH MWTD DVLD

CROD CROD

CPND_LANG ENG

RCO 0

HUNT

LHK 0

PLEV 02

PUID

DANI NO

AST

IAPG 0 *

AACS NO

ITNA NO

DGRP

MLWU_LANG 0

MLNG ENG

DNDR 0


CPND_LANG ROMAN

NAME Sigma 1140

XPLN 11

DISPLAY_FMT FIRST,LAST*

01 HOT U 116002 MARP 0 02

03

04

05

06

07

08

09

10

11

12

13

14

15

16

17 TRN

18 AO6

19 CFW 16

20 RGA

21 PRK

22 RNP

23 *

24 PRS

25 CHG

26 CPN

27

28

29

30

31

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5.2. Save Configuration

Expand Tools Backup and Restore on the left navigation panel and select Call Server.

Select Backup (not shown) and click Submit to save configuration changes as shown below.

The backup process will take several minutes to complete. Scroll to the bottom of the page to

verify the backup process completed successfully as shown below.

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6. Configuring Avaya Aura® Session Manager This section provides the procedures for configuring Session Manager. Session Manager is

configured via System Manager. The procedures include the following areas:

Log in to Avaya Aura® System Manager.

Administer SIP Domain.

Administer SIP Location.

Administer Adaptations.

Administer SIP Entities.

Administer Entity Links.

Administer Routing Policies.

Administer Dial Patterns.

It may not be necessary to create all the items above when creating a connection to the service

provider since some of these items would have already been defined as part of the initial Session

Manager installation. This includes items such as certain SIP domains, locations, SIP entities,

and Session Manager itself. However, each item should be reviewed to verify the configuration.

6.1. Log in to Avaya Aura® System Manager

Access the System Manager using a Web Browser by entering http://<FQDN >/SMGR, where

<FQDN> is the fully qualified domain name of System Manager. Log in using appropriate

credentials (not shown) and the Home tab will be presented with menu options shown below.

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Most of the configuration items are performed in the Routing Element. Click on Routing in the

Elements column shown above to bring up the Introduction to Network Routing Policy screen.

6.2. Administer SIP Domain

Create a SIP domain for each domain for which Session Manager will need to be aware in order

to route calls. Expand Elements Routing and select Domains from the left navigation menu,

click New (not shown). Enter the following values and use default values for remaining fields.

Name Enter a Domain Name. In the sample configuration, avaya.com was used.

Type Verify SIP is selected.

Notes Add a brief description [Optional].

Click Commit to save. The screen below shows the SIP Domain defined for the sample

configuration.

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6.3. Administer Locations

Locations can be used to identify logical and/or physical locations where SIP Entities reside for

purposes of bandwidth management and call admission control. To add a location, navigate to

Routing Locations in the left-hand navigation pane and click the New button in the right pane

(not shown). In the General section, enter the following values. Use default values for all

remaining fields:

Name: Enter a descriptive name for the location.

Notes: Add a brief description (optional).

The Location Pattern is used to identify call routing based on IP address. Session Manager

matches the IP address against the patterns defined in this section. If a call is from a SIP Entity

that does not match the IP address pattern then Session Manager uses the location administered

for the SIP Entity.

In the Location Pattern section, click Add and enter the following values.

IP Address Pattern Enter the logical pattern used to identify the location.

Notes Add a brief description [Optional].

Click Commit to save. The screenshot below shows the Location SM_7 defined for the

compliance testing.

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6.4. Administer Adaptations

Adaptations can be used to modify the called and calling party numbers to meet the requirements

of the service. The called party number present in the SIP INVITE Request URI is modified by

the Digit Conversion in the Adaptation. The example below was applied to the Avaya SBCE

SIP Entity and was used in test to convert numbers being passed between the Avaya SBCE and

Session Manager.

To add an adaptation, under the Routing tab select Adaptations on the left hand menu and then

click on the New button (not shown). Under Adaptation Details General:

In the Adaptation Name field enter an informative name.

In the Module Name field click on the down arrow and then select the <click to add

module> entry from the drop down list and type DigitConversionAdapter in the

resulting New Module Name field.

Module parameter MIME =no Strips MIME message bodies on egress from

Session Manager

fromto=true Modifies from and to headers of a message

Scroll down the page and under Digit Conversion for Incoming Calls to SM, click the Add

button and specify the digit manipulation to be performed as follows:

Enter the leading digits that will be matched in the Matching Pattern field.

In the Min and Max fields set the minimum and maximum digits allowed in the digit

string to be matched.

In the Delete Digits field enter the number of leading digits to be removed.

In the Insert Digits field specify the digits to be prefixed to the digit string.

In the Address to modify field specify the digits to manipulate by the adaptation. In this

configuration the dialed number is the target so both have been selected.

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This will ensure any incoming numbers matching +44 will have the + digit removed before being

presented to the Communication Server 1000.

In the Digit Conversion for Outgoing Calls to SM section, click Add and enter the following

values.

Enter the leading digits that will be matched in the Matching Pattern field.

In the Min and Max fields set the minimum and maximum digits allowed in the digit

string to be matched.

In the Delete Digits field enter the number of leading digits to be removed.

In the Insert Digits field specify the digits to be prefixed to the digit string.

In the Address to modify field specify the digits to manipulate by the adaptation. In this

configuration the dialed number is the target so both have been selected.

This will ensure any destination numbers beginning with 6 with have a specified CLID presented

on outbound calls.

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6.5. Administer SIP Entities

A SIP Entity must be added for each SIP-based telephony system supported by a SIP connection

to Session Manager. To add a SIP Entity, select SIP Entities on the left panel menu and then

click on the New button (not shown). The following will need to be entered for each SIP Entity.

Under General:

In the Name field enter an informative name

In the FQDN or IP Address field enter the IP address of Session Manager or the

signalling interface on the connecting system

In the Type field use Session Manager for a Session Manager SIP Entity, Other for a

Communication Server 1000 SIP Entity and SIP Trunk for the Avaya SBCE SIP Entity

In the Location field select the appropriate location from the drop down menu

In the Time Zone field enter the time zone for the SIP Entity

In this configuration there are three SIP Entities.

Session Manager SIP Entity

Communication Server 1000 SIP Entity

Avaya SBCE SIP Entity

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6.5.1. Avaya Aura® Session Manager SIP Entity

The following screens show the SIP entity for Session Manager. The FQDN or IP Address field

is set to the IP address of the Session Manager SIP signalling interface and Type is Session

Manager. Set the Location to that defined in Section 6.3 and the Time Zone to the appropriate

time.

Session Manager must be configured with the port numbers on the protocols that will be used by

the other SIP entities. To configure these scroll to the bottom of the page and under Port, click

Add, then edit the fields in the resulting new row.

In the Port field enter the port number on which the system listens for SIP requests.

In the Protocol field enter the transport protocol to be used for SIP requests.

In the Default Domain field, from the drop down menu select the domain added in

Section 6.2 as the default domain.

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6.5.1.1 Avaya Aura® Communication Server 1000 SIP Entity

The following screen shows the SIP entity for CS1000. The FQDN or IP Address field is set to

the IP address of the interface on CS1000 that will be providing SIP signalling and Type is

Other. Set the Location to that defined in Section 6.3 and the Time Zone to the appropriate

time.

Other parameters can be set for the SIP Entity as shown in the following screenshot, but for test,

these were left at default values.

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6.5.2. Avaya Session Border Controller for Enterprise SIP Entity

The following screen shows the SIP entity for the Avaya SBCE used for routing calls. The

FQDN or IP Address field is set to the IP address of the private interfaces administered in

Section 7 of this document. Set the location to that defined in Section 6.3, set Adaptation to one

created in Section 6.4 and the Time Zone to the appropriate time zone.

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6.6. Administer Entity Links

A SIP trunk between a Session Manager and another system is described by an Entity Link. To

add an Entity Link, select Entity Links on the left panel menu and click on the New button (not

shown). Fill in the following fields in the new row that is displayed.

In the Name field enter an informative name.

In the SIP Entity 1 field select Session Manager.

In the Protocol field enter the transport protocol to be used to send SIP requests.

In the Port field enter the port number to which the other system sends its SIP requests.

In the SIP Entity 2 field enter the other SIP Entity for this link, created in Section 6.4.

In the Port field enter the port number to which the other system expects to receive SIP

requests.

Select Trusted from the drop-down menu to make the other system trusted.

Click Commit to save changes. The following screenshot shows the Entity Links used in this

configuration.

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6.7. Administer Routing Policies

Routing policies must be created to direct how calls will be routed to a system. To add a routing

policy, select Routing Policies on the left panel menu and then click on the New button (not

shown).

Under General:

Enter an informative name in the Name field

Under SIP Entity as Destination, click Select, and then select the appropriate SIP entity

to which this routing policy applies

Under Time of Day, click Add, and then select the time range

The following screen shows the routing policy for CS1000.

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The following screen shows the Routing Policy for the Avaya SBCE.

6.8. Administer Dial Patterns

A dial pattern must be defined to direct calls to the appropriate telephony system. To configure a

dial pattern select Dial Patterns on the left panel menu and then click on the New button (not

shown).

Under General:

In the Pattern field enter a dialled number or prefix to be matched.

In the Min field enter the minimum length of the dialled number.

In the Max field enter the maximum length of the dialled number.

In the SIP Domain field select ALL or alternatively one of those configured in Section

6.2.

Under Originating Locations and Routing Policies:

Click Add, in the resulting screen (not shown).

Under Originating Location, select the location defined in Section 6.3 or ALL.

Under Routing Policies select one of the routing policies defined in Section 6.6.

Click Select button to save.

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The following screen shows an example dial pattern configured for the Avaya SBCE.

The following screen shows the test dial pattern configured for CS1000.

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7. Configure Avaya Session Border Controller for Enterprise This section describes the configuration of the Avaya Session Border Controller for Enterprise

(Avaya SBCE). The Avaya SBCE provides security and manipulation of signalling to provide an

interface to the Service Provider’s SIP Trunk that is standard where possible and adapted to the

Service Provider’s SIP implementation where necessary.

7.1. Access Avaya Session Border Controller for Enterprise

Access the Session Border Controller using a web browser by entering the URL https://<ip-

address>, where <ip-address> is the private IP address configured at installation. A log in

screen is presented.

Once logged in, a dashboard is presented with a menu on the left-hand side. The menu is used as

a starting point for all configuration of the Avaya SBCE.

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7.2. Define Network Management

Network information is required on the Avaya SBCE to allocate IP addresses and masks to the

interfaces. Note that only the A1 and B1 interfaces are used, typically the A1 interface is used for

the internal side and B1 is used for external. Each side of the Avaya SBCE can have only one

interface assigned.

To define the network information, navigate to Device Specific Settings Network

Management in the main menu on the left hand side and click on Add.

Enter details for the external interface in the dialogue box:

Enter a descriptive name in the Name field.

Enter the default gateway IP address for the external interface in the Default Gateway

field.

Enter the subnet mask in the Subnet Mask field.

Select the external interface to be used from the Interface drop down menu. In the test

environment, this was B1.

Click on Add and an additional row will appear allowing an IP address to be entered.

Enter the external IP address in the IP Address field and leave the Public IP and Gateway

Override fields blank.

Click on Finish to complete the interface definition.

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Click on Add to define the internal interface. Enter details in the dialogue box (not shown):

Enter a descriptive name in the Name field.

Enter the default gateway IP address for the internal interface in the Default Gateway

field.

Enter the subnet mask in the Subnet Mask field.

Select the internal interface to be used from the Interface drop down menu. In the test

environment, this was A1.

Click on Add and an additional row will appear allowing an IP address to be entered.

Enter the internal IP address in the IP Address field and leave the Public IP and Gateway

Override fields blank.

Click on Finish to complete the interface definition.

The following screenshot shows the completed Network Management configuration:

Select the Interface Configuration tab and click on Toggle State to enable the interfaces.

Note: to ensure that the Avaya SBCE uses the interfaces defined, the Application must be

restarted.

Click on System Management in the main menu (not shown).

Select Restart Application indicated by an icon in the status bar (not shown).

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7.3. Define Interfaces

When the IP addresses and masks are assigned to the interfaces, these are then configured as

signalling and media interfaces. Testing was carried out with TCP used for transport of

signalling between the Session Manager and the Avaya SBCE, and UDP for transport of

signalling between the Avaya SBCE and the BT Global Services SIP Trunk. This document

shows the configuration for TCP and UDP, if additional security is required, it’s recommended

to use TLS and port 5061.

7.3.1. Signalling Interfaces

To define the signalling interfaces on the Avaya SBCE, navigate to Device Specific Settings

Signaling Interface (not shown) in the main menu on the left hand side. Details of transport

protocol and ports for the external and internal SIP signalling are entered here.

Select Add and enter details of the external signalling interface in the pop-up menu.

In the Name field enter a descriptive name for the external signalling interface.

In the IP Address drop down menus, select the external network interface and IP

address. Note that when the external network interface is selected, the bottom drop down

menu is populated with the available IP addresses as defined in Section 7.2. In the test

environment, this was a single IP address 192.168.122.57.

Enter the UDP port number in the UDP Port field, 5060 is used for the BT Global

Services SIP Trunk.

The internal signalling interface is defined in the same way; the dialogue box is not shown:

Select Add and enter details of the internal signalling interface in the pop-up menu.

In the Name field enter a descriptive name for the internal signalling interface.

In the IP Address drop down menus, select the internal network interface and IP address.

Select TCP port number, 5060 is used for the Session Manager.

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The following screenshot shows details of the signalling interfaces:

Note. In the test environment, the internal IP address was 10.10.9.71.

7.3.2. Media Interfaces

To define the media interfaces on the Avaya SBCE, navigate to Device Specific Settings

Media Interface in the main menu on the left hand side. Details of the RTP and SRTP port

ranges for the internal and external media streams are entered here. The IP addresses for media

can be the same as those used for signalling.

Select Add and enter details of the external media interface in the pop-up menu.

In the Name field enter a descriptive name for the external media interface.

In the IP Address drop down menus, select the external network interface and IP

address. Note that when the external network interface is selected, the bottom drop down

menu is populated with the available IP addresses as defined in Section 7.2. In the test

environment, this was a single IP address 192.168.122.57.

Define the RTP Port Range for the media path with BT Global Services SIP Trunk,

during testing this was left at the default values.

The internal media interface is defined in the same way; the dialogue box is not shown:

Select Add and enter details of the internal media interface in the pop-up menu.

In the Name field enter a descriptive name for the internal media interface.

In the IP Address drop down menus, select the internal network interface and IP address.

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The following screenshot shows details of the media interfaces:

7.4. Define Server Interworking

Server interworking is defined for each server connected to the Avaya SBCE. In this case, BT

Global Services SIP Trunk is connected as the Trunk Server and the Session Manager is

connected as the Call Server.

To define server interworking on the Avaya SBCE, navigate to Global Profiles Server

Interworking in the main menu on the left hand side. To define Server Interworking for the

Session Manager, click on Add (not shown). A pop-up menu (not shown) is generated. In the

Name field enter a descriptive name for the Session Manager and click Next.

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Configuration of interworking includes Hold support, T.38 fax support and SIP extensions.

In the General dialogue box shown in the previous screenshot, check the T.38 Support

box. During testing, the rest of the parameters were left at default values.

Click on Next and Next again to go through the next two dialogue boxes. During testing,


In the final dialogue box, select None from the Extensions box. And click on Finish

To define Server Interworking for BT Global Services SIP Trunk, click on Add (not shown). A

pop-up menu (not shown) is generated. In the Name field enter a descriptive name for the BT

Global Services SIP Trunk and click Next.

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In the dialogue bow that appears, settings are as follows:

Check the Delayed SDP Handling box. This inserts an SDP into the empty INVITE sent

by the CS1000 when shuffling.

Check the T.38 box



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In the final dialogue box, select None from the Extensions box and click on Finish.

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7.5. Define Servers

A server definition is required for each server connected to the Avaya SBCE. In this case, BT

Global Services SIP Trunk is connected as the Trunk Server and the Session Manager is

connected as the Call Server. To define the BT Global Services SIP Trunk Server, navigate to

Global Profiles Server Configuration in the main menu on the left hand side. Click on Add

and enter an appropriate name in the pop-up menu (not shown). Click on Next and enter details

in the dialogue box.

In the Server Type drop down menu, select Trunk Server.

Click on Add to enter an IP address

In the IP Addresses / FQDN box, type the first BT Global Services network SBC

interface address.

In the Port box, enter the port to be used for the SIP Trunk. This was left blank during

testing which defaults to 5060 when UDP is used for transport.

In the Transport drop down menu, select UDP.

Click on Add and repeat the above for the alternative network SBC. Click on Next.



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Note: Although the Heartbeat configuration was left at default values for most of the testing, the

screenshot shows values used when verifying the SIP Trunk. For details, refer to Section 9.

The final dialogue box is the Advanced settings:

In the Interworking Profile drop down menu, select the Interworking Profile for the

BT Global Services SIP Trunk defined in Section 7.4.

Click Finish.

BT Global Services use two network SBCs for resilience. A separate Trunk Server configuration

is required for the alternative SBCs. Repeat the above process using the IP address of the

alternative SBC, in the test environment this was 192.168.221.23.

Use the process above to define the Call Server configuration for the Session Manager if not

already defined.

Ensure that Call Server is selected in the Server Type drop down menu in the General

dialogue box (not shown).

Ensure that the Interworking Profile defined for the Session Manager in Section 7.4 is

selected in the Interworking Profile drop down menu in the Advanced dialogue box (not

shown).

The following screenshot shows the completed entry for the Session Manager:

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7.6. Define Routing

Routing information is required for routing to BT Global SIP Trunk on the external side and the

Session Manager on the internal side. The IP addresses and ports defined here will be used as the

destination addresses for signalling.

To define routing to BT Global Service SIP Trunk, navigate to Global Profiles Routing in

the main menu on the left hand side. Click on Add and enter an appropriate name in the dialogue

box (not shown), click on Next and enter details for the Routing Profile:

In the Load Balancing drop down menu, select the method of load balancing required.

During testing this was set to Priority. If an even distribution across the network SBCs is

required, Round Robin could be used.

Click on Add to specify an IP address for the first network SBC.

Assign a priority in the Priority / Weight field

Select the Server Configuration defined in Section 7.5 in the Server Configuration drop

down menu. This automatically populates the Next Hop Address field

Repeat for the alternative network SBC. Click Finish.

Repeat the above process for the Routing Profile for the Session Manager:

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7.7. Topology Hiding

Topology hiding is used to hide local information such as private IP addresses and local domain

names. The local information can be overwritten with a domain name or IP addresses. The

default Replace Action is Auto, this replaces local information with IP addresses, generally the

next hop or external interfaces. Topology hiding has the advantage of presenting single Via and

Record-Route headers externally where multiple headers may be received from the enterprise,

particularly from the Session Manager. In some cases where Topology Hiding can’t be applied,

in particular the Contact header, IP addresses are translated to the Avaya SBCE external

addresses using NAT.

To define Topology Hiding for BT Global Service SIP Trunk, navigate to Global Profiles

Topology Hiding in the main menu on the left hand side. Click on Add and enter details in the

Topology Hiding Profile pop-up menu (not shown).

In the Profile Name field enter a descriptive name for BT Global Service SIP Trunk and

click Next.

Click on Add Header and select from the Header drop down menu.

Select IP or IP/Domain from the Criteria drop down menu depending on requirements.

Leave the Replace Action at the default value of Auto unless a specific domain name is

required. In this case, select Overwrite and define a domain name in the Overwrite

Value field.

Topology hiding was defined for all headers where the function is available.

To define Topology hiding for the Session Manager, follow the same process. This can be

simplified by cloning the profile defined for BT Global Service SIP Trunk. Do this by

highlighting the profile defined for the Session Manager and clicking on Clone.

Enter an appropriate name for the Session Manager and click on Next. Make any changes where

required, in the test environment the settings were left at the same values.

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7.8. Server Flows

Server Flows combine the previously defined profiles into two End Point Server Flows, one for

BT Global Services SIP Trunk and another for the Session Manager. This configuration ties all

the previously entered information together so that calls can be routed from the Session Manager

to BT Global Services SIP Trunk and vice versa.

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To define a Server Flow for the BT Global Services SIP Trunk, navigate to Device Specific

Settings End Point Flows.

Click on the Server Flows tab.

Select Add Flow and enter details in the pop-up menu.

In the Name field enter a descriptive name for the server flow for BT Global Services

SIP Trunk, in the test environment BT_Trunk was used.

In the Received Interface drop-down menu, select the internal SIP signalling interface

defined in Section 7.3. This is the interface that signalling bound for the BT SIP Trunk is

received on.

In the Signaling Interface drop-down menu, select the external SIP signalling interface

defined in Section 7.3. This is the interface that signalling bound for BT SIP Trunk is

sent on.

In the Media Interface drop-down menu, select the external media interface defined in

Section 7.3. This is the interface that media bound for BT SIP Trunk is sent on.

In the Routing Profile drop-down menu, select the routing profile of the Session

Manager defined in Section 7.6.

In the Topology Hiding Profile drop-down menu, select the topology hiding profile of

the BT SIP Trunk defined in Section 7.7 and click Finish.

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To define a Server Flow for the Session Manager, navigate to Device Specific Settings End

Point Flows.

Click on the Server Flows tab.

Select Add Flow and enter details in the pop-up menu.

In the Name field enter a descriptive name for the server flow for the Session Manager,

in the test environment CPE was used.

In the Received Interface drop-down menu, select the external SIP signalling interface

defined in Section 7.3. This is the interface that signalling bound for the Session Manager

is received on.

In the Signaling Interface drop-down menu, select the internal SIP signalling interface

defined in Section 7.3. This is the interface that signalling bound for the Session Manager

is sent on.

In the Media Interface drop-down menu, select the internal media interface defined in

Section 7.3. This is the interface that media bound for the Session Manager is sent on.

In the Routing Profile drop-down menu, select the routing profile of BT SIP Trunk

defined in Section 7.6.

In the Topology Hiding Profile drop-down menu, select the topology hiding profile of

the Session Manager defined in Section 7.7 and click Finish.

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The information for all Server Flows is shown on a single screen on the Avaya SBCE.

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8. Configure BT SIP Trunk Equipment The configuration of the BT Global Services equipment used to support the SIP Trunk is outside

the scope of these Application Notes and will not be covered. To obtain further information on

BT Global Services equipment and system configuration please contact an authorised BT

representative.

9. Verification Steps This section provides verification steps that may be performed in the field to verify that the

solution is configured properly.

9.1. Avaya Communication Server 1000 Verification

This section illustrates sample verifications that may be performed using the Avaya CS1000

Element Manager GUI.

9.1.1. IP Network Maintenance and Reports Commands

From Element Manager, navigate to System IP Network Maintenance and Reports as

shown below. In the resultant screen on the right, click the Gen CMD button.

The General Commands page is displayed. A variety of commands are available by selecting

an appropriate Group and Command from the drop-down menus, and selecting Run.

To check the status of the SIP Gateway to Session Manager in the sample configuration, select

Sip from the Group menu and SIPGwShow from the Command menu. Click Run. The example

output below shows that Session Manager has SIPNPM Status “Active”.

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The following screen shows a means to view registered SIP telephones. The screen shows the

output of the Command sigSetShowAll in Group SipLine.

The following screen shows a means to view IP UNIStim telephones. The screen shows the

output of the Command isetShow in Group Iset.

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9.2. Verify Avaya Communication Server 1000 Operational Status

Expand System on the left navigation panel and select Maintenance. Select LD 96 - D-Channel

from the Select by Overlay table and the D-Channel Diagnostics function from the Select by

Functionality table as shown below.

Select Status for D-Channel (STAT DCH) command and click Submit to verify status of

virtual D-Channel as shown below. Verify the status of the following fields.

APPL_STATUS Verify status is OPER

LINK_STATUS Verify status is EST ACTV

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9.3. Verify Avaya Aura® Session Manager Operational Status

9.3.1. Verify Avaya Aura® Session Manager is Operational

Navigate to Elements Session Manager Dashboard (not shown) to verify the overall

system status for Session Manager. Specifically, verify the status of the following fields as

shown below.

Navigate to Elements Session Manager System Status Security Module Status (not

shown) to view more detailed status information on the status of Security Module for the specific

Session Manager. Verify the Status column displays Up as shown below.

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9.3.2. Verify SIP Entity Link Status

Navigate to Elements Session Manager System Status SIP Entity Monitoring (not

shown) to view more detailed status information for one of the SIP Entity Links. Select the SIP

Entity for CS1000 from the All Monitored SIP Entities table (not shown) to open the SIP

Entity, Entity Link Connection Status page.

Verify the status of the SIP link is up between Session Manager and CS1000 by going through

the same process as outlined above but selecting the SIP Entity for the Avaya SBCE in the All

Monitored SIP Entities: table.

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9.3.3. Verify Avaya Aura® Session Manager Instance

The creation of a Session Manager Instance provides the linkage between System Manager and

Session Manager. This was most likely done as part of the initial Session Manager installation.

To add a Session Manager, navigate to Elements Session Manager Session Manager

Administration in the left-hand navigation pane and click on the New button in the right pane

(not shown). If Session Manager instance already exists, click View (not shown) to view the

configuration. Enter/verify the data as described below and shown in the following screen:

In the General section, enter the following values:

SIP Entity Name: Select the SIP Entity created for Session

Manager

Description: Add a brief description (optional)

Management Access Point Host Name/IP: Enter the IP address of Session Manager

management interface

The following screen shows Session Manager values used for the compliance test.

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In the Security Module section, enter the following values:

SIP Entity IP Address: Should be filled in automatically based on the SIP Entity

Name. Otherwise, enter IP address of Session Manager

signaling interface

Network Mask: Enter the network mask corresponding to the IP address of

Session Manager

Default Gateway: Enter the IP address of the default gateway for Session

Manager

Use default values for the remaining fields. Click Save (not shown). The following screen shows

the remaining Session Manager values used for the compliance test.

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9.4. Avaya Session Boarder Controller for Enterprise Verification

This section contains verification steps that may be performed using the Avaya Session Border

Controller for Enterprise.

9.4.1. Incidents

The Incidents Log Viewer display alerts captured by the Avaya SBCE. Select the Incidents link

along the top of the screen.

The following screen shows example SIP messages that do not match a Server Flow for an

incoming message.

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9.4.2. Trace Settings

The Trace Settings tool is for configuring and displaying call traces and packet captures for the

Avaya SBCE.

To define the trace, navigate to Device Specific Settings Advanced Options

Troubleshooting Trace in the main menu on the left hand side and select the Packet

Capture tab.

Select the SIP Trunk interface from the Interface drop down menu

Select the signalling interface IP address from the Local Address drop down menu

Enter the IP address of the network SBC in the Remote Address field or enter a * to

capture all traffic

Specify the Maximum Number of Packets to Capture, 10000 is shown as an example

Specify the filename of the resultant pcap file in the Capture Filename field

To view the trace, select the Captures tab and click on the relevant filename in the list of traces.

The trace is viewed as a standard pcap file in Wireshark. If the SIP trunk is working correctly, a

SIP response in the form of a 200 OK will be seen from the BT Global Services network.

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10. Conclusion These Application Notes describe the configuration necessary to connect Avaya Aura®

Communication Server R7.65, Avaya Aura® Session Manager R7.0 and Avaya Session Border

Controller for Enterprise R7.0 to BT Global Services SIP Trunk. BT Global Services SIP Trunk

is a SIP-based Voice over IP solution providing businesses a flexible, cost-saving alternative to

traditional hardwired telephony trunks. The service was successfully tested with a number of

observations listed in Section 2.2. At the time of writing, an ongoing issue remains with loss of

media after 30 minutes on long duration calls. This is under investigation.

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11. Additional References This section references the documentation relevant to these Application Notes. Additional Avaya

product documentation is available at http://support.avaya.com.

[1] Migrating and Installing Avaya Appliance Virtualization Platform, Release 7.0, Nov

2015.

[2] Upgrading and Migrating Avaya Aura® applications to 7.0, Release 7.0, Nov 2015.

[3] Deploying Avaya Aura® applications, Release 7.0, Oct 2015

[4] Deploying Avaya Aura® System Manager Release 7.0 Nov 2015

[5] Upgrading Avaya Aura® System Manager to Release 7.0, Nov 2015.

[6] Administering Avaya Aura® System Manager for Release 7.0 Release 7.0, Nov 2015

[7] Deploying Avaya Aura® Session Manager on VMware , Release 7.0 August 2015

[8] Upgrading Avaya Aura® Session Manager Release 7.0, August 2015

[9] Administering Avaya Aura® Session Manager Release 7.0, August 2015

[10] Avaya Communication Server 1000 Installation and Commissioning, Document

Number NN43041-310

[11] Linux Platform Base and Applications Installation and Commissioning Avaya

Communication Server 1000, Document Number NN43001-315

[12] Software Input Output Reference – Maintenance Avaya Communication Server 1000,

Document Number NN43001-711

[13] Deploying Avaya Session Border Controller for Enterprise, Release 7.0, August 2015

[14] Upgrading Avaya Session Border Controller for Enterprise, Release 7.0, August 2015

[15] Administering Avaya Session Border Controller for Enterprise, Release 7.0, Nov 2015

[16] RFC 3261 SIP: Session Initiation Protocol, http://www.ietf.org/

http://support.avaya.com/

http://www.ietf.org/

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Appendix A – Communication Server 1000 Software

Communication Server 1000 call server patches and plug ins TID: 46379

VERSION 4121

System type is - Communication Server 1000/CPPM Linux

CPPM - Pentium M 1.4 GHz

IPMGs Registered: 1

IPMGs Unregistered: 0

IPMGs Configured/unregistered: 0

RELEASE 7

ISSUE 65 P +

IDLE_SET_DISPLAY NORTEL

DepList 1: core Issue: 01(created: 2015-09-28 04:19:50 (est))

MDP>LAST SUCCESSFUL MDP REFRESH :2015-11-12 14:50:17(Local Time)

MDP>USING DEPLIST ZIP FILE DOWNLOADED :2013-09-28 04:30:29(est)

SYSTEM HAS NO USER SELECTED PEPS IN-SERVICE

LOADWARE VERSION: PSWV 100+

INSTALLED LOADWARE PEPS : 1

PAT# CR # PATCH REF # NAME DATE FILENAME

00 wi01057886 ISS1:1OF1 DSP2AB07 13/09/2013 DSP2AB07.LW

ENABLED PLUGINS : 2

PLUGIN STATUS PRS/CR_NUM MPLR_NUM DESCRIPTION

-------------------------------------------------------------

201 ENABLED Q00424053 MPLR08139 PI:Cant XFER OUTG TRK TO OUTG TRK

501 ENABLED Q02138637 MPLR30070 Enables blind transfer to a SIP endpoint even

if SIP UPDATE is not supported by the far en

Communication Server 1000 call server deplists

VERSION 4121

RELEASE 7

ISSUE 65 P +

DepList 1: core Issue: 01 (created: 2013-05-28 04:19:50 (est))

IN-SERVICE PEPS

PAT# CR # PATCH REF # NAME DATE FILENAME SPECINS

000 wi01058359 ISS1:1OF1 p32331_1 16/11/2015 p32331_1.cpl NO

001 wi01064599 iss1:1of1 p32580_1 16/11/2015 p32580_1.cpl NO

002 wi01056067 ISS1:1OF1 p32457_1 16/11/2015 p32457_1.cpl NO

003 wi01063263 ISS1:1OF1 p32573_1 16/11/2015 p32573_1.cpl NO

004 wi01065842 ISS1:1OF1 p32478_1 16/11/2015 p32478_1.cpl NO

005 wi01062607 ISS1:1OF1 p32503_1 16/11/2015 p32503_1.cpl NO

006 wi01070756 ISS1:1OF1 p32444_1 16/11/2015 p32444_1.cpl NO

007 wi01039280 ISS1:1OF1 p32423_1 16/11/2015 p32423_1.cpl NO

008 wi01087543 ISS1:1OF1 p32662_1 16/11/2015 p32662_1.cpl NO

009 wi00933195 ISS1:1OF1 p32491_1 16/11/2015 p32491_1.cpl NO

010 wi01071379 ISS1:1OF1 p32522_1 16/11/2015 p32522_1.cpl NO

011 wi01068669 ISS1:1OF1 p32333_1 16/11/2015 p32333_1.cpl NO

012 wi01066991 ISS1:1OF1 p32449_1 16/11/2015 p32449_1.cpl NO

013 wi01070474 iss1:1of1 p32407_1 16/11/2015 p32407_1.cpl NO

014 WI0110261 ISS1:1OF1 p32758_1 16/11/2015 p32758_1.cpl NO

015 wi01094305 ISS1:1OF1 p32640_1 16/11/2015 p32640_1.cpl NO

016 wi01047890 ISS1:1OF1 p32697_1 16/11/2015 p32697_1.cpl NO

017 wi01055300 ISS1:1OF1 p32543_1 16/11/2015 p32543_1.cpl NO

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018 wi01082456 ISS1:1OF1 p32596_1 16/11/2015 p32596_1.cpl NO

019 wi01058621 ISS1:1OF1 p32339_1 16/11/2015 p32339_1.cpl NO

020 wi01061484 ISS1:1OF1 p32576_1 16/11/2015 p32576_1.cpl NO

021 wi01078723 ISS1:1OF1 p32532_1 16/11/2015 p32532_1.cpl NO

022 wi01048457 ISS1:1OF1 p32581_1 16/11/2015 p32581_1.cpl NO

023 wi01075355 ISS1:1OF1 p32594_1 16/11/2015 p32594_1.cpl NO

024 wi01053597 ISS1:1OF1 p32304_1 16/11/2015 p32304_1.cpl NO

025 wi01045058 ISS1:1OF1 p32214_1 16/11/2015 p32214_1.cpl NO

026 wi01075359 ISS1:1OF1 p32671_1 16/11/2015 p32671_1.cpl NO

027 wi01025156 ISS1:1OF1 p32136_1 16/11/2015 p32136_1.cpl NO

028 wi01061481 ISS1:1OF1 p32382_1 16/11/2015 p32382_1.cpl NO

029 wi01035976 ISS1:1OF1 p32173_1 16/11/2015 p32173_1.cpl NO

030 wi01088775 ISS1:1OF1 p32659_1 16/11/2015 p32659_1.cpl NO

031 wi01070465 iss1:1of1 p32562_1 16/11/2015 p32562_1.cpl NO

032 wi01088585 ISS1:1OF1 p32656_1 16/11/2015 p32656_1.cpl NO

033 wi01063864 ISS1:1OF1 p32410_1 16/11/2015 p32410_1.cpl YES

034 wi01034961 ISS1:1OF1 p32144_1 16/11/2015 p32144_1.cpl NO

035 wi01055480 ISS1:1OF1 p32712_1 16/11/2015 p32712_1.cpl NO

036 wi01034307 ISS1:1OF1 p32615_1 16/11/2015 p32615_1.cpl NO

037 wi01065118 ISS1:1OF1 p32397_1 16/11/2015 p32397_1.cpl NO

038 wi01075360 iss1:1of1 p32602_1 16/11/2015 p32602_1.cpl NO

039 wi00884716 ISS1:1OF1 p32517_1 16/11/2015 p32517_1.cpl NO

040 wi01068851 ISS1:1OF1 p32439_1 16/11/2015 p32439_1.cpl NO

041 wi01053314 ISS1:1OF1 p32555_1 16/11/2015 p32555_1.cpl NO

042 wi01059388 iss1:1of1 p32628_1 16/11/2015 p32628_1.cpl NO

043 wi01087528 ISS1:1OF1 p32700_1 16/11/2015 p32700_1.cpl NO

044 wi01072027 ISS1:1OF1 p32689_1 16/11/2015 p32689_1.cpl NO

045 wi01052428 ISS1:1OF1 p32606_1 16/11/2015 p32606_1.cpl NO

046 wi01053920 ISS1:1OF1 p32303_1 16/11/2015 p32303_1.cpl NO

047 wi01070468 iss1:1of1 p32418_1 16/11/2015 p32418_1.cpl NO


049 wi01060826 ISS1:1OF1 p32379_1 16/11/2015 p32379_1.cpl NO

050 wi01075352 ISS1:1OF1 p32603_1 16/11/2015 p32603_1.cpl NO

051 wi01043367 ISS1:1OF1 p32232_1 16/11/2015 p32232_1.cpl NO

052 wi01083584 ISS1:1OF1 p32619_1 16/11/2015 p32619_1.cpl NO

053 wi01060241 ISS1:1OF1 p32381_1 16/11/2015 p32381_1.cpl NO

054 wi01053195 ISS1:1OF1 p32297_1 16/11/2015 p32297_1.cpl NO

055 wi00897254 ISS1:1OF1 p31127_1 16/11/2015 p31127_1.cpl NO

056 wi01061483 ISS1:1OF1 p32359_1 16/11/2015 p32359_1.cpl NO

057 wi01085855 ISS1:1OF1 p32658_1 16/11/2015 p32658_1.cpl NO

058 wi01075353 ISS1:1OF1 p32613_1 16/11/2015 p32613_1.cpl NO

059 wi01070471 ISS1:1OF1 p32415_1 16/11/2015 p32415_1.cpl NO

060 wi01074003 ISS1:1OF1 p32421_1 16/11/2015 p32421_1.cpl NO

061 wi01060382 iss1:1of1 p32623_1 16/11/2015 p32623_1.cpl YES

062 wi01068042 ISS1:1OF1 p32669_1 16/11/2015 p32669_1.cpl NO


064 wi01065922 ISS1:1OF1 p32516_1 16/11/2015 p32516_1.cpl NO

065 wi01057403 ISS1:1OF1 p32591_1 16/11/2015 p32591_1.cpl NO

066 wi01069441 ISS1:1OF1 p32097_1 16/11/2015 p32097_1.cpl NO

067 wi01070473 ISS1:1OF1 p32413_1 16/11/2015 p32413_1.cpl NO

068 wi01056633 ISS1:1OF1 p32322_1 16/11/2015 p32322_1.cpl NO

069 wi01052968 ISS1:1OF1 p32540_1 16/11/2015 p32540_1.cpl NO

070 wi01072032 ISS1:1OF1 p32448_1 16/11/2015 p32448_1.cpl NO

071 wi01073100 ISS1:1OF1 p32599_1 16/11/2015 p32599_1.cpl NO

072 wi01035980 ISS1:1OF1 p32558_1 16/11/2015 p32558_1.cpl NO

073 wi01041453 ISS1:1OF1 p32587_1 16/11/2015 p32587_1.cpl NO

074 wi01032756 ISS1:1OF1 p32673_1 16/11/2015 p32673_1.cpl NO

075 wi01092300 ISS1:1OF1 p32692_1 16/11/2015 p32692_1.cpl NO

076 wi00996734 ISS1:1OF1 p32550_1 16/11/2015 p32550_1.cpl NO

077 wi01022599 ISS1:1OF1 p32080_1 16/11/2015 p32080_1.cpl NO

078 wi01060341 ISS1:1OF1 p32578_1 16/11/2015 p32578_1.cpl NO

079 wi01091447 ISS1:1OF1 p32675_1 16/11/2015 p32675_1.cpl NO

080 wi01070580 ISS1:1OF1 p32380_1 16/11/2015 p32380_1.cpl NO

081 wi01089519 ISS1:1OF1 p32665_1 16/11/2015 p32665_1.cpl NO

082 WI01077073 ISS1:1OF1 p32534_1 16/11/2015 p32534_1.cpl NO

083 wi01080753 ISS1:1OF1 p32518_1 16/11/2015 p32518_1.cpl NO

084 wi01065125 ISS1:1OF1 p32416_1 16/11/2015 p32416_1.cpl NO

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Communication Server 1000 signaling server service updates

In System service updates: 41

PATCH# IN_SERVICE DATE SPECINS REMOVABLE NAME

0 Yes 14/07/14 YES YES cs1000-csmWeb-7.65.16.22-2.i386.000

1 Yes 14/10/15 YES YES cs1000-dmWeb-7.65.16.23-4.i386.000

3 Yes 15/10/15 NO YES cs1000-sps-7.65.16.23-1.i386.000

4 Yes 14/07/14 YES YES cs1000-patchWeb-7.65.16.22-4.i386.000

5 Yes 14/10/15 YES YES cs1000-linuxbase-7.65.16.23-19.i386.000

7 Yes 14/07/14 YES YES cs1000-csoneksvrmgr-7.65.16.22-5.i386.000

8 Yes 27/09/13 NO YES cs1000-pd-7.65.16.21-00.i386.000

9 Yes 27/09/13 NO YES cs1000-shared-carrdtct-7.65.16.21-

01.i386.000

10 Yes 27/09/13 NO YES cs1000-shared-tpselect-7.65.16.21-

01.i386.000

11 Yes 14/07/14 YES YES cs1000-baseWeb-7.65.16.22-4.i386.000

12 Yes 27/09/13 NO yes cs1000-dbcom-7.65.16.21-00.i386.000

16 Yes 14/10/15 NO YES cs1000-Jboss-Quantum-7.65.16.23-5.i386.000

17 Yes 15/10/15 YES YES cs1000-cs-7.65.P.100-03.i386.000

18 Yes 15/10/15 NO YES bash-3.2-33.el5_11.4.i386.000

19 Yes 15/10/15 YES YES cs1000-shared-pbx-7.65.16.23-1.i386.000

20 Yes 15/10/15 YES YES cs1000-emWeb_6-0-7.65.16.23-3.i386.000

21 Yes 15/10/15 NO YES libxml2-2.6.26-2.1.25.el5_11.i386.000

22 Yes 15/10/15 NO YES libxml2-python-2.6.26-

2.1.25.el5_11.i386.000

23 Yes 02/04/14 NO YES cs1000-shared-omm-7.65.16.21-2.i386.000

24 Yes 15/10/15 NO YES freetype-2.2.1-32.el5_9.1.i386.000

26 Yes 15/10/15 NO YES cs1000-cs1000WebService_6-0-7.65.16.23-

1.i386.000

27 Yes 14/07/14 YES YES cs1000-oam-logging-7.65.16.22-4.i386.000

28 Yes 15/10/15 YES YES cs1000-ftrpkg-7.65.16.23-1.i386.000

29 Yes 15/10/15 NO YES cs1000-cppmUtil-7.65.16.23-4.i686.000

30 Yes 02/10/13 NO YES cs1000-snmp-7.65.16.21-00.i686.000

31 Yes 14/07/14 YES YES cs1000-csv-7.65.16.22-2.i386.000

33 Yes 14/07/14 YES YES cs1000-nrsm-7.65.16.22-3.i386.000

34 Yes 14/07/14 YES YES cs1000-mscTone-7.65.16.22-2.i386.000

35 Yes 14/07/14 YES YES cs1000-mscMusc-7.65.16.22-4.i386.000

36 Yes 14/07/14 YES YES cs1000-mscConf-7.65.16.22-2.i386.000

38 Yes 02/04/14 YES YES cs1000-emWebLocal_6-0-7.65.16.22-1.i386.000

39 Yes 15/10/15 NO YES tzdata-2015a-1.el5.i386.000

40 Yes 02/04/14 YES YES cs1000-ipsec-7.65.16.22-1.i386.000

41 Yes 15/10/15 YES YES cs1000-tps-7.65.16.23-15.i386.000

43 Yes 15/10/15 YES YES kernel-2.6.18-406.el5.i686.000

44 Yes 15/10/15 YES YES cs1000-vtrk-7.65.16.23-76.i386.000

45 Yes 15/10/15 YES YES cs1000-bcc-7.65.16.23-10.i386.000

47 Yes 14/07/14 YES YES cs1000-mscAnnc-7.65.16.22-2.i386.000

48 Yes 14/07/14 YES YES cs1000-mscAttn-7.65.16.22-2.i386.000

49 Yes 14/07/14 NO YES cs1000-gk-7.65.16.22-1.i386.000

53 Yes 14/07/14 YES YES cs1000-shared-xmsg-7.65.16.22-1.i386.000

CMN; Reviewed:

RRR m/d/y



73 of 74

BTGS_CS1K76_SM

Communication Server 1000 system software

Product Release: 7.65.16.00

Base Applications

base 7.65.16 [patched]

NTAFS 7.65.16

sm 7.65.16

cs1000-Auth 7.65.16

Jboss-Quantum n/a [patched]

cnd 7.65.16

lhmonitor 7.65.16

baseAppUtils 7.65.16

dfoTools 7.65.16

cppmUtil n/a [patched]

oam-logging n/a [patched]

dmWeb n/a [patched]

baseWeb n/a [patched]

ipsec n/a [patched]

Snmp-Daemon-TrapLib n/a [patched]

ISECSH 7.65.16

patchWeb n/a [patched]

EmCentralLogic 7.65.16

Application configuration: CS+SS+NRS+EM

Packages:

CS+SS+NRS+EM

Configuration version: 7.65.16-00

cs 7.65.16 [patched]

dbcom 7.65.16.21 [patched]

cslogin 7.65.16

sigServerShare 7.65.16 [patched]

csv 7.65.16 [patched]

tps 7.65.16 [patched]

vtrk 7.65.16 [patched]

pd 7.65.16.21 [patched]

sps 7.65.16 [patched]

ncs 7.65.16

gk 7.65.16 [patched]

nrsm 7.65.16 [patched]

nrsmWebService 7.65.16

managedElementWebService 7.65.16

EmConfig 7.65.16

emWeb_6-0 7.65.16 [patched]

emWebLocal_6-0 7.65.16 [patched]

csmWeb 7.65.16 [patched]

bcc 7.65.16 [patched]

ftrpkg 7.65.16 [patched]

cs1000WebService_6-0 7.65.16 [patched]

mscAnnc 7.65.16 [patched]

mscAttn 7.65.16 [patched]

mscConf 7.65.16 [patched]

mscMusc 7.65.16 [patched]

mscTone 7.65.16 [patched]

CMN; Reviewed:

RRR m/d/y



74 of 74

BTGS_CS1K76_SM


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are the property of their respective owners. The information provided in these Application

Notes is subject to change without notice. The configurations, technical data, and

recommendations provided in these Application Notes are believed to be accurate and

dependable, but are presented without express or implied warranty. Users are responsible for

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